Abstract: A computer-implemented method and system are provided for profiling or analyzing audio data based on an audio parameter. The audio data is divided into audio data into segments. Each segment is characterized by the audio parameter in a defined range. Effects are selected according to the segments meeting criteria for the audio parameter of the effects. A media presentation is authored or generated using the selected effects.

Abstract: According to one embodiment, an audio processor includes a detector and a controller. The detector is configured to detect a first interval during which a state in which a reproduced sound by an audio signal is assumed to be silent continues for at least a first time. The controller is configured to change, during the first interval, an output level of the audio signal.

Abstract: Embodiments described herein provide for detecting movement in proximity to a playback device and responsively modifying the audio that is output from one or more playback devices. According to an embodiment, a user is able to wave his or her hand over a playback device to modify the audio that is output from the playback device, one or more different playback devices, or all of the playback devices. The embodiments are particularly useful as a snooze feature, in which an alarm from a playback device (or group of playback devices) is snoozed for a time when triggered by the user's movement.

Abstract: A system for generating one or more enhanced audio signals such that one or more sound levels corresponding with sounds received from one or more sources of sound within an environment may be dynamically adjusted based on contextual information is described. The one or more enhanced audio signals may be generated by a head-mounted display device (HMD) worn by an end user within the environment and outputted to earphones associated with the HMD such that the end user may listen to the one or more enhanced audio signals in real-time. In some cases, each of the one or more sources of sound may correspond with a priority level. The priority level may be dynamically assigned depending on whether the end user of the HMD is focusing on a particular source of sound or has specified a predetermined level of importance corresponding with the particular source of sound.

Abstract: An audio output controlling apparatus and method, and a mobile terminal system using the same control settings of an audio out port according to whether an audio frame includes an audio signal. The audio output controlling method includes receiving an audio frame, determining whether an audio signal is included in the audio frame, and setting, depending on whether the audio signal is included in the audio frame, a power voltage provided to an audio out port. The mobile terminal includes a wireless transceiver; an audio unit, through an audio out port, an audio frame among the wireless data; and an audio output controlling unit to control a power voltage of the audio out port. The apparatus includes a section determining unit; and a power controlling unit to set a power voltage provided to an audio out port according to whether the audio signal is included in the audio frame.

Abstract: Techniques are provided for enhancing the audio component of streamed media provided by an application on a computing device, referred to collectively herein as an enhance audio mode. The mode may be configured to enhance audio by adjusting the volume gain during playback of streamed media from a specific application (e.g., Netflix or Pandora). In some cases, detection of playback from a specific media streaming application may be based on detection of a system file update, which may occur, for example, in a user space level. In some cases, the volume gain adjustment may be applied by an audio codec included with the computing device, which may occur, for example, in a kernel space level. The volume gain adjustment may be determined by analyzing the amplitude of the audio component of media streamed from a specific application relative to the speaker(s) included with the computing device.

Abstract: A variable gain amplifier amplifies an input audio signal. A first control unit monitors the level of the input audio signal, and sets the gain of the variable gain amplifier to a predetermined fixed value when this level is lower than a predetermined minimum threshold. A second control unit monitors the level of an output signal VO of the variable gain amplifier. When this level exceeds a predetermined maximum threshold, the second control unit lowers the gain of the variable gain amplifier. When this level becomes smaller than this threshold, it raises the gain. When the level of the input audio signal is smaller than the minimum threshold, the first control unit disables the gain control operation of the variable gain amplifier controlled by the second control unit. A sub-amplifier amplifies the input audio signal. An output mixing unit mixes signals output from the variable gain amplifier and the sub-amplifier.

Abstract: A technique is provided for automatically adjusting the volume, or magnitude, of an audio signal. The technique includes calculating an average power associated with a segment of an input audio signal, determining whether the average power is greater than an estimated signal level associated with one or more previously-processed segments of the input audio signal and, depending on the determination, either calculating an updated estimated signal level by subtracting from the average power an attenuated difference between the estimated signal level and the average power or setting the updated estimated signal level to the average power. A gain to be applied to the segment of the input audio signal is then determined based on the updated estimated signal level and a target signal level for an output audio signal.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed. The present invention includes receiving the audio signal and preset information; obtaining preset matrix from the preset information, wherein the preset matrix indicates contribution degree of the object to output channel; and adjusting output level of the object by using the preset matrix. Accordingly, without user's setting for each object, if preset metadata to be applied to an audio signal is selected with reference to previously-set preset metadata, levels of objects included in the audio signal can be easily adjusted using preset rendering data corresponding to the selected preset metadata.

Abstract: An amplifier circuit and a method of amplification using automatic gain control (AGC) are disclosed. A method for reducing distortions incurred by an audio signal when being rendered by an electronic device is described. The method comprises receiving an input signal; determining signal strength; determining a frequency-dependent AGC filter; wherein the frequency-dependent AGC filter is adapted to selectively attenuate the input signal within a number N of predetermined frequency ranges, according to corresponding N degrees of attenuation; wherein N predetermined frequency ranges depend upon a rendering characteristic of the electronic device; and wherein the N-degrees of attenuation depend upon the signal strength; and attenuating the input signal using the frequency-dependent AGC filter to obtain an output signal for rendering by the electronic device.

Abstract: Systems and methods are described which facilitate quick and accurate extraction of the true noise level from a noise signal that includes additional signals, such as speech, in a cost effective implementation. Aspects of the invention allow the use of one microphone to simultaneously detect background noise as well as speech, while avoiding problems associated with artificially high background noise indication due to inclusion of the speech component in the noise determination. Additionally, systems and methods are described for altering system gain based on accurate noise level determinations.

Abstract: A method and system is presented for sampling analog signals in a manner that avoids the effects of signal clipping due to a limited dynamic range. A method and device for sampling an analog input using multiple gains, or gain mask, is described. By using different gains during different time quanta, a subset of the sampled points may effectively be attenuated before being sampled and converted to a digital representation. If clipping occurs during the sampling process, the true values of the clipped samples may be interpolated using the amplitudes of the non-clipped samples, which may not have been attenuated. Such interpolation may include constructing and/or solving a constraint optimization problem using linear programming. In one embodiment, such a problem may be constructed and/or solved by using sign information from the clipped samples and/or by imposing a sparsity assumption on the signals during the reconstruction process.

Abstract: Automatic gain control systems disclosed herein can incorporate a confidence metric that can estimate the accuracy of gain adjustments calculated by an automatic gain control module. The confidence metric may be based on a percentage of valid audio samples in a given period of time. Based on the confidence metric, the AGC response may be reduced, delayed, frozen, or otherwise altered from the baseline gain adjustment. Time-averaging process may be used to estimate the input signal power level and determine an appropriate baseline gain adjustment. Additionally, weighting functions can be adjusted to prevent overestimation of the signal power.

Abstract: Techniques pertaining to providing an appropriate recording mode to achieve optimum record data in accordance with the recording environment are described. In one embodiment, a current audio recording mode is initially selected to be one of a low sensitivity audio recording mode and a high sensitivity audio recording mode. A digital gain for digital audio recording data in the current audio recording mode is obtained. The current audio recording mode is automatically switched between the low sensitivity audio recording mode and the high sensitivity audio recording mode when the digital gain reaches a gain threshold.

Abstract: Provided is an audio processing apparatus including a frequency domain conversion unit configured to convert an audio signal input from a microphone to a frequency domain for each of frames, and a gain adjustment unit configured to perform gain adjustment for each of bands on the audio signal converted to the frequency domain. The gain adjustment unit acquires an autocorrelation value of power of the audio signal between the frames for each of the bands, and sets an adjustment amount of the gain in accordance with the acquired autocorrelation value.

Abstract: A method and an apparatus process an audio signal. The method of processing an audio signal includes: separating the audio signal into a first harmonic signal and a first percussion signal; adjusting gains of the first harmonic signal and the first percussion signal to generate a second harmonic signal and a second percussion signal; and adding the second harmonic signal and the second percussion signal. The method can generally improve the apparatus for controlling a gain and adopt a reproducing volume level to a current environment by separating the audio signal into a harmonic signal and a percussion signal to adjust a gain. The audio signal can have a high output level without non-linear distortion.

Abstract: A device (1) is arranged for controlling the sound levels of a group of audio channels including a user selected main channel (MC) and at least one auxiliary channel (AC1; AC2). The audio channels can be rendered simultaneously. The device comprises automatic level adjustment means (12, 13) for adjusting the sound level of the at least one auxiliary channel relative to the main channel. The level adjustment means (12, 13) may be arranged for adapting the respective sound levels to the content or signal characteristics of each associated audio channel.

Abstract: An audio signal amplitude adjusting device and method that can always adjust the amplitude of an input audio signal to such a level as to be easy to hear with making it follow each shift in the level of the input audio signal, thus preventing the occurrence of the sense of aural discomfort and an interruption of a voice. The gain of a variable attenuator for adjusting the amplitude of the audio signal is controlled in various ways. When a momentary large noise sound is contained in the audio signal, the gain reduced state invoked to suppress the effect of the noise sound is detected, and thereafter if the amplitude of the audio signal has become smaller than the reference level due to the noise sound ending, the gain is increased each period that is shortest among the periods of control employed.

Abstract: A computerized method adjusts two presented audio signals using an audio player. A first audio signal is received and played through an audio playing unit. When a second audio signal that is inserted to be played is received in time during the playing of the first audio signal, a peak amplitude X of a volume of the first audio signal and a peak amplitude Y of a volume of the second audio signal are extracted. Each of the peak amplitude X and the peak amplitude Y are compared with a predetermined threshold value A. The first audio signal and/or the second audio signal are adjusted according to the comparison, and the first audio signal and/or the second audio signal is outputted to the audio playing unit and played through the audio playing unit according to the adjustment for the first audio signal and/or the second audio signal.

Abstract: A system and method for providing layered audio which allows a user to selectively and simultaneously listen to multiple audio sources for limited time periods and switch between these layered audio sources. The system allows a user to discernibly listen to two content sources at the same time.

Abstract: Systems and method for adapting a gain of an audio signal containing at least two different tracks with different signal level ranges. An example system includes a loudness determination unit for determining a perceived loudness of the audio input signal based on a psychoacoustic model of a human hearing. The system further includes a loudness adaptation unit configured to determine a loudness adaptation indicator based on the perceived loudness and a predetermined loudness, and to introduce the loudness adaptation indicator into the audio signal to generate an adapted gain signal. A gain determination unit is configured to adapt the gain for outputting the audio signal containing the at least two tracks based on the loudness adaptation indicators in the adapted gain signal at about the predetermined loudness.

Abstract: An application for a television with a headphone dock. The headphone dock provides status to the television as to when a set of headphones are docked (idle) or undocked (in use). The television changes routing of audio signals based upon the status of the headphones.

Abstract: An audio interruption and buffering playback system includes a primary audio source for reproducing primary audio content and a secondary audio source for reproducing secondary audio content. A processing device detects an interruption event that includes the secondary audio source reproducing secondary audio content having priority over the primary audio source. The processor mutes the output of the primary audio content in response to the interruption event. A buffer buffers the audio content from the primary audio source during the interruption event. The buffered audio content is reproduced by the processor to the user at an accelerated playback speed following the interruption event.

Abstract: A multimedia controller including a processor, the controller configured to: provide a user interface for a player group, wherein the player group includes a plurality of players in a local area network, and wherein each player is configured to playback a multimedia output from a multimedia source; accept an input to facilitate formation of the player group, indicating that at least two of the players in the local area network are to be included in the player group; for each of the plurality of players within the player group, accept an input to adjust a volume associated with the player, that causes the player to adjust its volume; and accept an input to adjust a volume associated with the player group, wherein the input to adjust the volume associated with the group causes the players in the player group to adjust their volumes.

Abstract: It is described a method for controlling an adaptation of a behavior of an audio device (100) to a current acoustic environmental condition. The method comprises (a) monitoring an audio output signal (x(t), x?(t)) being provided to an acoustic output device (110) of the audio device (100) for outputting an acoustic output signal, (b) measuring an audio input signal (z(t)) being provided by an acoustic input device (120) of the audio device (100), wherein the audio input signal (z(t)) is indicative for a feedback portion of the acoustic output signal and for the current acoustic environmental condition, (c) determining a relation between the audio output signal (x?(t)) and the audio input signal (z(t)) and (d) adapting the behavior of the audio device (100) based on the determined relation.

Abstract: The sound device includes an audio-information output unit, an analysis unit, an audio-division-spectrum output unit, a noise-division-spectrum output unit and a correction unit. The analysis unit receives audio information from the audio-information output unit, and then outputs sound spectrum information. The noise-division-spectrum output unit outputs sound-volume information for each critical band width of a noise, and the audio-division-spectrum output unit outputs the sound-volume information for each critical band width of the sound-spectrum information. The correction unit corrects the information from the audio-division-spectrum output unit based on the information from the noise-division-spectrum output unit. The audio-signal properties can be well corrected corresponding to the auditory-sense properties of the human, and thus the audio sound, in which an uncomfortable feeling to the auditory sense of the human has been adequately controlled, can be transmitted to a user.

Abstract: A method and apparatus for automatic gain control of an amplifier in which the method and apparatus detects excessive increase in the sound volume or temperature rise in the audio system and reduces the gain of the amplifier by various means. Each gain control circuit has a comparator to compare a signal with threshold values and an integrator to integrate output signals of the comparator with respect to a predetermined time length to reduce a gain of the amplifier. The method and apparatus conducts dynamic suppressing of excess signals and thereby controlling output level of the amplifier at multiple stages without immediately shutting down the audio system. In the case where the sound volume increases to a degree that may damage the amplifier or speaker, a shutdown signal is generated to shutdown the operation of the whole system.

Abstract: An audio system includes a level controller receiving an input audio signal from an audio medium. The level controller includes a gain adjust module producing an output audio signal based on the input audio signal. A filter arrangement is communicatively coupled to the level controller. The filter arrangement receives the output audio signal and divides the output signal into a plurality of filtered signals. Each of the filtered signals corresponds to a respective frequency band. A multiband dynamics controller is communicatively coupled to the filter arrangement. The multiband dynamics controller receives the filtered signals and applies a respective gain characteristic to each of the filtered signals to thereby produce a plurality of gain adjust signals. The gain adjust signals are combined into a multiband dynamics controller output signal.

Abstract: A display device and a method of driving the display, wherein the volume of background music is controlled depending upon the color or brightness of display images, and the power consumption is reduced. The display device includes a display unit with a plurality of pixels, a signal controller, and a volume controller. The display unit displays images by selectively activating pixels from among the plurality of pixels in accordance with input video signals and input image control signals. The signal controller reads the input video signals to generate on-pixel signals having information about the ratio of a number of the active pixels compared to a total number of pixels in the plurality of pixels per each frame.

Abstract: One aspect of the present invention can include a playback device capable of externally receiving a playback signal for playing back a sound or an image, and capable of transmitting the playback signal to the playback device if it is detected that a predetermined electric source was externally supplied.

Abstract: Techniques for optimizing a player based on the addition of a second player are disclosed. In an embodiment, when a first player no longer needs to play certain audio frequencies due to the addition of a second player, the gain of the first player is automatically increased as part of the setup process. In another embodiment, when a first player needs to play certain audio frequencies, for example due to the removal of a second player, the gain of the first player is automatically decreased. Many other embodiments are disclosed.

Abstract: An apparatus and method for outputting audio signals of a portable communication device are provided. The apparatus includes a plurality of audio tables including different volume control values or tone control values, and allows a user to selectively set audio tables corresponding to each of audio paths equipped with the portable communication device. Further, the apparatus analyzes a surrounding environment during a voice call, determines the absence or presence of noise, and suitably controls a volume or a tone of audio signals based on the result of the determination before outputting the audio signals. Accordingly, the apparatus can provide an improved calling condition in various calling environments.

Abstract: A processing method and a processing apparatus for processing an output audio signal from an audio capture device. The output audio signal can be processed by a first processing module in a manner such that a preliminary signal and a first stage processed signal can be derived. The processing method includes providing power estimation signals, providing at least one cross power estimation signal, providing a leakage approximation and providing a control signal estimation.

Abstract: A display device and a driving method thereof where different sounds are output depending upon characteristics of display images. The display device includes a display unit having a plurality of pixels, a signal controller, and a sound controller. The display unit displays images by selectively activating pixels from among the plurality of pixels in accordance with input video signals and input image control signals. The signal controller reads the input video signals to generate on-pixel signals having information about a ratio of a number of the activated pixels compared to a total number of pixels in the plurality of pixels per each frame and generating a sound output start signal.

Abstract: The invention provides a method and device for enhancing the listening qualities of an audio file by providing the listener with a plurality of modified equalized audio files. Each modified equalized audio file having a consistent loudness level but different audio characteristics. Hence, for an input audio file the current invention allows the listener to individually select the best audio characteristics for them to listen to the content of the input audio file according to their particular requirements without them needing to adjust the loudness level in playback. The invention further enables the listener to switch between the multiple equalized audio files during playback. The invention further includes a SN detector and reducer to eliminate the adverse effects of the presence of sudden, strong noise in the input audio file in the process of generating the plurality of modified equalized audio files.

Abstract: A volume control system comprises: an amplifying apparatus; and a controller connectable to the amplifying apparatus. The amplifying apparatus includes; a first receiving section for receiving a volume value from the controller, a first input section into which an instruction for changing the volume value is input by a user's operation, a first transmitting section for transmitting the volume value input into the first input section to the controller, and a volume control section for, when the first receiving section receives the volume value or the changed volume value is input into the first input section, controlling a volume value of the amplifying apparatus based on the volume value.

Abstract: A sound reproducing device has a loudspeaker arranged to produce sound from an audio signal provided by an audio signal source. A microphone is positioned to pick up ambient noise and generate a microphone signal which comprises the noise. An ambient noise cancellation (ANC) system receives the microphone signal from the microphone and generates anti-noise corresponding to the ambient noise in the microphone signal. An automatic polarity adaptation (AAP) system monitors the ANC system and, when a decision criterion is fulfilled, causes a switch in polarity for the generated anti-noise.

Abstract: A coefficient calculation unit calculates a coefficient (p) according to input signals (x[n]). A balance setting unit decides, according to this coefficient (p), a coefficient used to control the level of a signal in a digital level correction unit and a coefficient used to adjust the characteristics of a signal in a characteristic correction unit, thus allowing the digital level correction unit and characteristic correction unit to execute adaptive processing.

Abstract: Disclosed are a mobile terminal and a control method thereof. The mobile terminal includes: a sound output module configured to output a sound; a user input unit configured to sense a control command for controlling a volume level of the output sound; and a controller configured to control a volume level of the output sound based on the sensed control command, configured to determine whether the controlled volume level corresponds to a preset volume level, and if the controlled volume level corresponds to the preset volume level, configured to output a signal corresponding to the preset volume level in the form of a signal different from the output sound.

Abstract: A method is disclosed for rendering audio on a computing device. The method is performed by one or more processors of the computing device. The one or more processors determine at least a position or an orientation of the computing device based on one or more inputs detected by one or more sensors of the computing device. The one or more processors control the output level of individual speakers in a set of two or more speakers based, at least in part, on the at least determined position or orientation of the computing device.

Abstract: A speaker array control method includes steps of detecting a position of an audience located in front of a speaker array, wherein the speaker array includes N speakers and N is a positive integer larger than one; defining a target and a non-target with respect to an i-th speaker of the N speakers according to the position of the audience, wherein i is a positive integer smaller than or equal to N; calculating a weighting vector for the i-th speaker according to the target and the non-target; adjusting a directionality of an output signal of the i-th speaker by the weighting vector and reducing energy of a plurality of side lobes of the output signal of the i-th speaker; and controlling the i-th speaker to output the adjusted output signal when the energy of each of the side lobes is smaller than a threshold.

Abstract: A method of adjusting a loudness of an audio signal in real time may include receiving an electronic audio signal and dividing the audio signal into a plurality of frames. Processing of the frames may be performed in real time. The processing may include measuring initial loudness values for blocks of samples in a frame to produce a plurality of initial loudness values, and computing a weighted average of at least some of the initial loudness values. The weights may be selected based on one or more of the recency of the initial loudness values, variation of the initial loudness values, and estimated information content of the audio signal. The processing may further include selectively discarding at least some of the loudness values that reach an adaptive loudness threshold. Weights can be adjusted based on the variation of the loudness values of the audio signal.

Abstract: A method of adjusting a loudness of an audio signal in real time may include receiving an electronic audio signal and dividing the audio signal into a plurality of frames. Processing of the frames may be performed in real time. The processing may include measuring initial loudness values for blocks of samples in a frame to produce a plurality of initial loudness values, and computing a weighted average of at least some of the initial loudness values. The weights may be selected based on one or more of the recency of the initial loudness values, variation of the initial loudness values, and estimated information content of the audio signal. The processing may further include selectively discarding at least some of the loudness values that reach an adaptive loudness threshold. Weights can be adjusted based on the variation of the loudness values of the audio signal.

Abstract: Optimal microphone volumes are automatically set for computer applications based on determination of peak volume levels and noise levels from one or more digital audio captures. The peak volume levels and noise levels can be advantageously determined based on distribution curves of sample volume levels in the digital audio captures. Clipping can be automatically compensated for by estimating peak unclipped capture volume levels from the distribution curves.

Abstract: A range finding audio system automatically modifies the audio output of an audio source based on the distance of a listener from the speakers. A speaker in an audio system may include a range device coupled with a controller. The range device may utilize infrared, laser, or acoustic technology to determine the distance between the speaker and the listener. The controller may transfer distance information to an audio interface of a processor unit. The audio lo interface may include a positioning routine to modify the audio output according to the distance from the speaker to the listener. Alternatively, the controller may perform the functions ascribed to the positioning routine making the necessary modifications to the audio output based on the distance information.

Abstract: A technique is disclosed to implement crossfading of audio tracks. In one embodiment, the function describing the fade out of the ending audio track and/or the slope describing the fade in of the beginning audio track may be altered to increase the perceptible overlap of the two tracks. In another embodiment, the duration of the fade out and/or of the fade in may be altered to increase the perceptible overlap of the two tracks. In other embodiments, one or both of the function and/or duration of the fade out and/or fade in effect may be altered to improve the perceptibility of the overlap or the audio tracks.

Abstract: An apparatus is provided for remotely monitoring a plurality of amplifiers on a display device. The amplifiers are provided to process audio signals and directly or indirectly connected to a network. In the apparatus, a collecting section collects a group code from each of the amplifiers through the network. The group code is allocated to each amplifier according to a grouping system for grouping the plurality of the amplifiers. A display control section displays a tree diagram of the amplifiers according to the collected group codes on the display device as a graphical representation of the grouping system.

Abstract: An amplification apparatus comprises: an amplifier circuit that amplifies an input audio signal and outputs an output audio signal; a first signal level determining unit that determines whether or not a signal level of the output audio signal outputted from the amplifier circuit is lower than a silence determination reference value; and a power-supply control unit that shifts the amplification apparatus from a power-on state to a power-off state, when it is determined that the signal level of the output audio signal outputted from the amplifier circuit is lower than the silence determination reference value.

Abstract: This audio signal output device includes an audio signal output portion, a volume setting information acquisition portion, and a control portion acquiring maximum volume information of volume information of an audio signal of content. The control portion is configured to calculate the maximum output volume of an external device on the basis of the maximum volume information of the content and volume setting information of the external device and perform control of lowering the output volume of the external device when the maximum output volume is higher than a prescribed volume threshold.

Abstract: A sound volume correcting device includes: a variable gain unit controlling a gain of an input audio signal on the basis of a gain control signal; a voice average level detector detecting an average level of a human voice signal in the input audio signal; and a gain control signal generator generating the gain control signal for controlling the gain of the input audio signal using the average level of the human voice signal detected by the voice average level detector as a reference level and supplying the generated gain control signal to the variable gain unit.